After setting up cannot receive or make calls

FreePBX Installation / Upgrade
1

I’ve completed the nerdvittles.com tutorials, installed a X-lite softphone, and subscribed and completed all required steps outlined by the voip provider. FreePBX is setup on my network (ISO from PBXinaFlash) and the computer is able to ping internet sites. The softphone shows a correct login with the PBX. I have even placed the computer in the DMZ of my router to minimize firewall issues, but I still cannot make or receive calls.

Calling Out) When I try calling out from the softphone, the message states that the number dialed is no longer in service. The Asterisk log states:
NoOp(“SIP/192.168.0.163-096905d8”, "Received incoming SIP connection from unknown peer to ") in new stack
[Feb 4 21:36:30] VERBOSE[7520] logger.c: – Executing [@from-sip-external:2] Set(“SIP/192.168.0.163-096905d8”, “DID=”) in new stack
[Feb 4 21:36:30] VERBOSE[7520] logger.c: – Executing [@from-sip-external:3] Goto(“SIP/192.168.0.163-096905d8”, “s|1”) in new stack
[Feb 4 21:36:30] VERBOSE[7520] logger.c: – Goto (from-sip-external,s,1)
[Feb 4 21:36:30] VERBOSE[7520] logger.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/192.168.0.163-096905d8”, “0?from-trunk||1”) in new stack
[Feb 4 21:36:30] VERBOSE[7520] logger.c: – Executing [s@from-sip-external:2] Set(“SIP/192.168.0.163-096905d8”, “TIMEOUT(absolute)=15”) in new stack
[Feb 4 21:36:30] VERBOSE[7520] logger.c: – Channel will hangup at 2008-02-05 04:36:45 UTC.

Incoming Calls) The service provider logs state that the call is answered and the message states that the call cannot be completed as dialed.

I have modified the /etc/asterisk/sip.conf to identify my DynDNS account and NAT, but not in any other places. Any suggestions as to what else I should configure?

2

for starters, your phone is not configured properly, it is not registered with the server which is what the error message above is about.

3

Another issue is that you have edited the sip.conf file.

Look in the file and you should see this:
; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.

One thing to learn with FreePBX is that you really only edit the files that end in _custom.conf and if you find any referance to a file being automaticly generated, etc it will get overwritten when changes or upgrades happen. Yes in your case you are not working with the jitterbuffer but the sip.conf file will get replaced from time to time and you’ll loose you changes if they are not placed into of those files.

If I remember correctly settings for your provider should go in sip_registrations_custom.conf

4